I set: sip debug peer 3000 (my test extension) and dialed #3643873 Here is the output:
<-- SIP read from 192.168.1.59:17456: INVITE sip:%233643...@192.168.2.10 <sip%3a%25233643...@192.168.2.10> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.59:17456 ;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:3...@192.168.1.59:17456> To: "#3643873"<sip:%233643...@192.168.2.10 <sip%3a%25233643...@192.168.2.10> > From: "Test"<sip:3...@192.168.2.10 <sip%3a3...@192.168.2.10>>;tag=76126b35 Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="3000",realm="asterisk",nonce="6a7a2c99",uri=" sip:%233643...@192.168.2.10 <sip%3a%25233643...@192.168.2.10> ",response="7bcf9339c154ef939bd575aeaaef1860",algorithm=MD5 User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 317 v=0 o=- 8 2 IN IP4 192.168.1.59 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.59 t=0 0 m=audio 34194 RTP/AVP 107 0 8 101 a=alt:1 2 : MsRCET/S fNqrHReN 192.168.200.113 34194 a=alt:2 1 : pf8wX3Si UdjtGUj2 192.168.1.59 34194 a=fmtp:101 0-15 a=rtpmap:107 BV32/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (13 headers 12 lines)--- Using INVITE request as basis request - NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE. Sending to 192.168.1.59 : 17456 (non-NAT) Found user '3000' Found RTP audio format 107 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.59:34194 Found description format BV32 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for %233643873 in from-internal (domain 192.168.2.10) Reliably Transmitting (no NAT) to 192.168.1.59:17456: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.59:17456 ;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport;received=192.168.1.59 From: "Test"<sip:3...@192.168.2.10 <sip%3a3...@192.168.2.10>>;tag=76126b35 To: "#3643873"<sip:%233643...@192.168.2.10 <sip%3a%25233643...@192.168.2.10> >;tag=as3d020428 Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: <sip:%233643...@192.168.2.10 <sip%3a%25233643...@192.168.2.10>> Content-Length: 0 --- aikphone*CLI> <-- SIP read from 192.168.1.59:17456: ACK sip:%233643...@192.168.2.10 <sip%3a%25233643...@192.168.2.10> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.59:17456 ;branch=z9hG4bK-d8754z-e210db3acd1ed62b-1---d8754z-;rport To: "#3643873"<sip:%233643...@192.168.2.10 <sip%3a%25233643...@192.168.2.10> >;tag=as3d020428 From: "Hull Barrett"<sip:3...@192.168.2.10 <sip%3a3...@192.168.2.10> >;tag=76126b35 Call-ID: NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE. CSeq: 2 ACK Content-Length: 0 --- (7 headers 0 lines)--- Destroying call 'NWYxNDZmM2VjOTc0ZDM3OWE5ZDU4N2MxMDAzOTc0YmE.' On Wed, May 5, 2010 at 12:40 PM, Danny Nicholas <da...@debsinc.com> wrote: > Ok. I’m confused. I was interpreting what you wrote to say that you are > doing this: > > 1. pick up sip phone attached to pbx1 (1.2 box) > 2. dial #5551212 > 3. command dial(iax2/trunk/5551212,30,r) gets executed on 1.2 box > 4. 1.4 box should fall into _XXXXXXX and do DAHDI dial? > > > > If this is correct, where is the IAX command in your CLI output. > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel > *Sent:* Wednesday, May 05, 2010 10:11 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Hash Dial Pattern Problems > > > > I am on the 1.2 box and see nothing with the verbose cranked up. I do see > the following when tailing the asterisk full log during the calls: > > May 5 11:09:46 DEBUG[26538] chan_sip.c: Setting NAT on RTP to 0 > > May 5 11:09:46 DEBUG[26538] chan_sip.c: Checking SIP call limits for > device 3000 > > May 5 11:09:46 DEBUG[26538] chan_sip.c: Stopping retransmission on > 'Njg3MjI5N2IzNDk3NWYxZTMzMzFmMzEwNzc2ZDE1NTE.' of Response 2: Match Found > > > > > > On Wed, May 5, 2010 at 10:39 AM, Danny Nicholas <da...@debsinc.com> wrote: > > Ok – you have to be getting something or you wouldn’t get that message. > You are looking at CLI on the 1.2 or 1.4 box? If you’re looking at the 1.4 > side, you won’t see anything until a connection is made (although you should > see some kind of credential reject or something??) > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel > *Sent:* Wednesday, May 05, 2010 9:31 AM > > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Hash Dial Pattern Problems > > > > Nothing..goes directly to "The person you are calling is unavailable". > > On Wed, May 5, 2010 at 9:46 AM, Danny Nicholas <da...@debsinc.com> wrote: > > Set verbose to 5 and see if you get a CLI output. > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel > *Sent:* Wednesday, May 05, 2010 8:39 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Hash Dial Pattern Problems > > > > I am sorry it is not 1.6 but 1.4.30 (argh.. to many boxes) > > The other box is running 1.2.1 > > Thanks, > > David > > On Wed, May 5, 2010 at 9:28 AM, Danny Nicholas <da...@debsinc.com> wrote: > > Which 1.6 are you running? I dropped my 1.6.1.6 back to 1.4.30 because my > other 2 1.4.30 boxes wouldn’t talk to it properly. > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Nickel > *Sent:* Wednesday, May 05, 2010 8:23 AM > *To:* asterisk-users@lists.digium.com > *Subject:* [asterisk-users] Hash Dial Pattern Problems > > > > I have two Asterisk boxe. One is running 1.6 and the other 1.2 > > The users on the 1.2 system press # plus a local 7 digit number to place > local calls through the trunk to the 1.6 box. > > For some reason this dial pattern fails right away with "unavailable". > There is no activity in the CLI. Other patterns for the trunk work just > fine. > > Dial pattern: > #|. or #|NXXXXXX > > exten => _#.,1,Dial(IAX2/trunk/${EXTEN:1},30,r) > exten => _#.,2,Congestion > > I have been beating my end with the problem for three days. Any suggestions > would be much appreciated. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? 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